VideoSinks receive the new kind of VideoFrames and will replace
VideoRenderers. Converting from old texture frames to VideoFrames will
involve conversion to I420 so it is not recommended to use VideoSinks
before all sources produce VideoFrames.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/3002553002
Cr-Commit-Position: refs/heads/master@{#19335}
Relanding after adding "androidnetworkmonitor_jni.h" header to jni/
directory, since some clients were including it directly.
This CL breaks peerconnection_jni.cc apart, into one file for each
class. It also moves the methods for converting between C++/Java
structs into "java_native_conversion.cc", and uses a consistent naming
scheme ("JavaToNativeX, NativeToJavaX"). These files go into a new
"pc" directory, of which deadbeef@ is added as an owner.
It also moves some relevant files to the "pc" directory that belong
there: ownedfactoryandthreads, androidnetworkmonitor_jni, and
rtcstatscollectorcallbackwrapper. This directory is intended to hold
all the files that deal with the PeerConnection API specifically, or
related classes (like DataChannel, RtpSender, MediaStreamTrack) that
are tied to it closely.
BUG=webrtc:8055
Review-Url: https://codereview.webrtc.org/2992103002
Cr-Commit-Position: refs/heads/master@{#19241}
Reason for revert:
Borken in the internal projects.
Original issue's description:
> Break peerconnection_jni.cc into multiple files, in "pc" directory.
>
> This CL breaks peerconnection_jni.cc apart, into one file for each
> class. It also moves the methods for converting between C++/Java
> structs into "java_native_conversion.cc", and uses a consistent naming
> scheme ("JavaToNativeX, NativeToJavaX"). These files go into a new
> "pc" directory, of which deadbeef@ is added as an owner.
>
> It also moves some relevant files to the "pc" directory that belong
> there: ownedfactoryandthreads, androidnetworkmonitor_jni, and
> rtcstatscollectorcallbackwrapper. This directory is intended to hold
> all the files that deal with the PeerConnection API specifically, or
> related classes (like DataChannel, RtpSender, MediaStreamTrack) that
> are tied to it closely.
>
> deadbeef@webrtc.org is added as an owner of the new "pc" subdirectory.
>
> BUG=webrtc:8055
>
> Review-Url: https://codereview.webrtc.org/2992103002
> Cr-Commit-Position: refs/heads/master@{#19223}
> Committed: dd7d8f1b60TBR=magjed@webrtc.org,sakal@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8055
Review-Url: https://codereview.webrtc.org/2989323002
Cr-Commit-Position: refs/heads/master@{#19226}
This CL breaks peerconnection_jni.cc apart, into one file for each
class. It also moves the methods for converting between C++/Java
structs into "java_native_conversion.cc", and uses a consistent naming
scheme ("JavaToNativeX, NativeToJavaX"). These files go into a new
"pc" directory, of which deadbeef@ is added as an owner.
It also moves some relevant files to the "pc" directory that belong
there: ownedfactoryandthreads, androidnetworkmonitor_jni, and
rtcstatscollectorcallbackwrapper. This directory is intended to hold
all the files that deal with the PeerConnection API specifically, or
related classes (like DataChannel, RtpSender, MediaStreamTrack) that
are tied to it closely.
deadbeef@webrtc.org is added as an owner of the new "pc" subdirectory.
BUG=webrtc:8055
Review-Url: https://codereview.webrtc.org/2992103002
Cr-Commit-Position: refs/heads/master@{#19223}
Previously, the matrix in VideoFrame was used to crop and scale the
frame. This caused complications because webrtc::VideoFrame doesn't
include a matrix. cropAndScale method is added to VideoBuffer class for
cropping and scaling instead.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/2990583002
Cr-Commit-Position: refs/heads/master@{#19179}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
Will reland in two different commits to preserve git blame history.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
This change also wires up the rest of the production code in
webrtc/sdk/android to be built when the directory is a dependency.
BUG=webrtc:7613
NOTRY=True
Change-Id: Ideda181970a5a570c3f8148b033e471e926243d1
Reviewed-on: https://chromium-review.googlesource.com/548038
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18791}
The implementation creates an Android hardware video decoder. It is built
around the same patterns as the HardwareVideoEncoderFactory.
This change pulls some shared code and constants into a common "utils" class.
Finally, adds an instrumentation test for the HardwareVideoDecoder.
BUG=webrtc:7760
Change-Id: Iea6eaae7727925743cb54f7c3153a6c07d62f55d
Reviewed-on: https://chromium-review.googlesource.com/536254
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18686}
This CL makes the WebRTC Java Wrapper more modular and allows the android
users to build WebRTC without audio and video(DataChannel only).
The BUILD file in sdk/android/ is modified to support modular WebRTC.
The peerconnection_jni.cc is split into peerconnection_jni.cc, video_jni.cc,
video_renderer_jni.cc and ownedfactoryandthreads.h/cc.
Add new modular build targets to JNI layer: audio_jni, video_jni,
null_audio_jni, null_video_jni. The users can link with different
targets to for different WebRTC functionalities.
This is split from CL: https://codereview.webrtc.org/2854123003/TBR=magjed@webrtc.org
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2939203002
Cr-Commit-Position: refs/heads/master@{#18647}
Adds the VideoEncoderFactory interface and implements it for use with HardwareVideoEncoder. This uses MediaCodecVideoEncoder's initialization code as an example.
BUG=webrtc:7760
Change-Id: I9fbc93ce9ac4ad866750a4386c4f15e800a3073e
Reviewed-on: https://chromium-review.googlesource.com/530063
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18636}
This functionality is needed when sending C++ I420 buffers to Java
VideoSinks or Java encoders.
Bug: webrtc:7749
Change-Id: Ied783470b90b9d2e0cb5930795f35de4a296d499
Reviewed-on: https://chromium-review.googlesource.com/532961
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18597}
This is to make it possible to override the rtc_task_queue target only.
BUG=none
Review-Url: https://codereview.webrtc.org/2931273002
Cr-Commit-Position: refs/heads/master@{#18534}
These interfaces will be used by the future refactoring that will
allow clients to provide custom codec implementations.
Change-Id: If199bc2807e1c27094c05983c62fa43d2eec5700
Bug: webrtc:7760
Reviewed-on: https://chromium-review.googlesource.com/522065
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18441}
This new VideoFrame class closesly matches the C++ webrtc::VideoFrame
and webrtc::VideoFrameBuffer classes. It's supposed to replace the
existing VideoRenderer.I420Frame. The purpose is to clean up the code
and support more frame formats.
BUG=webrtc:7749
Review-Url: https://codereview.webrtc.org/2915083002
Cr-Commit-Position: refs/heads/master@{#18404}
Very similar to the current interface, but matches the new C++ structure, and
exposes the stats values as Objects which can be downcast to more specific
types (where the previous API only exposed the values as strings).
BUG=webrtc:6871
Review-Url: https://codereview.webrtc.org/2807933003
Cr-Commit-Position: refs/heads/master@{#17746}
This class has been deprecated for a long time and has been replaced by
Camera1Capturer.
BUG=webrtc:7440
Review-Url: https://codereview.webrtc.org/2789183004
Cr-Commit-Position: refs/heads/master@{#17538}
Moves CameraCapturer, CameraSession, Camera1Session and Camera2Session
away from the public API.
BUG=webrtc:7172
Review-Url: https://codereview.webrtc.org/2699713004
Cr-Commit-Position: refs/heads/master@{#16723}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.
Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
> needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:
ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
Move file capturer/renderer tests from the AppRTCMobile tests directory
to the WebRTC tests directory. These tests do not test AppRTCMobile but
rather WebRTC functionality. Therefore, they belong in WebRTC tests
directory.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2632233002
Cr-Commit-Position: refs/heads/master@{#16115}
